Prior Art Products Information
Network Phones
Networks such as IP/Ethernet, T1 and ISDN can connect telephony terminals to one another. Voice and video data streams are carried on these networks per their respective protocols. Telephony devices such as SIP phones are network based equivalents to the conventional telephone. Telephony Gateways provide an interface between phones on a local data network such as a local area network (LAN) and a conventional telephony network such as analog phone lines, digital telephone interfaces such as T1 and ISDN commonly found in the PBX.
Mediatrics PSTN Gateway
This device provides a SIP to PSTN interface. It terminates a SIP call on Ethernet IP side and relays it on PSTN side.
Intel Audio PBX Gateway
This device provides a SIP to PBX interface. It terminates a SIP call on Ethernet IP side and relays it on PBX side.
AudioCodes PRI Gateway
This device provides a SIP to PBX interface. It terminates a SIP call on Ethernet IP side and relays it on PBX side.
CISCO SIP Phone
Pingtel Xpressa SIP Phone
These phones are connected to an Ethernet network instead of dedicated telephone lines. They send and receive a single audio data stream and require a media conferencing unit (MCU) to connect 3 or more telephones into the same call.
H.320/H323 Phones
These devices use the H.320/H.323 protocol instead of the SIP protocol to establish connections. They also require the use of an MCU to conference 3 or more phones.
Present Invention
The conventional phone terminals connect to a central processing device that receives the audio and video streams from the terminals, mixes the audio and formats the incoming video. The results are sent back to the terminal for display and playback.
The ViPr™ system differentiates itself from other conferencing systems in that in a conference call of 3 or more terminals each terminal sends audio and video data to the other terminals using multicast IP or point-to-multipoint (PMP) ATM and each terminal receives the audio and video data streams from each of the other terminals and processes them accordingly. The incoming audio streams were encoded using standardized algorithms by the transmitting terminals. The terminals receiving these audio streams decode and add the incoming audio streams. The blended audio stream is played out of a speaker system for the user to hear.
The distributed video and audio processing that ViPr uses is incompatible with the MCU architecture that the SIP phones and H devices use. In order to interface to these devices, a means and apparatus called the Unicast Audio Mixer (UAM) was developed.
ViPr terminals communicate to the UAM as if it was another ViPr terminal without video. The unicast devices such as SIP phones communicate with the UAM as if it was another SIP phone. This function of presenting one interface to the ViPr terminals and another to the unicast devices is called a back-to-back user agent (B2BUA). The UAM is the only device that can implement a B2BUA between ViPr terminals and unicast devices.
The UAM handles a large number of simultaneous conferences involving one or more unicast devices and one or more ViPr terminals.
Another feature of the UAM is to act as a bridge between two different networks. For example, if all the ViPr terminals are on an ATM network and the unicast devices are on an Ethernet network then the UAM would be configured with one ATM NIC and one Ethernet NIC. Along with the SIP signaling and audio processing, the UAM would bridge these two networks.
In order to support this functionality, the ViPr system includes a UAM that adds seamless conferencing functionality between the ViPr terminals and telephone users (i.e. PSTN, Mobile phones and SIP phones) by converting an upstream unicast telephone audio stream to point-to-multipoint audio streams (i.e. PMP-SVC or IP Multicast) and mixing/converting downstream PMP/multicast ViPr audio streams to unicast telephone audio streams as well as performing downstream audio transcoding of ViPr audio from the wideband 16 bit/16 KHz PCM encoding to G.711 or G.722.
An additional functionality provided by the UAM is that of an Intermedia gateway that converts IP/UDP audio streams to ATM SVC audio streams and vice-versa. This functionality enables the interoperability between ViPr systems deployed in ATM environments and SIP-based Voice-over-IP (VoIP) telephony gateways on Ethernet networks.
The current ViPr system provides support for telephony systems through SIP-based analog and digital telephony gateways. This functionality enables ViPr users to make/receive point-to-point calls to/from telephone users. However, they do not allow a ViPr user to add a telephone call to a ViPr conference. This is due to the unicast nature of telephone calls and the inability of the telephony gateways to convert them to PMP/multicast streams. The ViPr UAM will enhance the ViPr system's support for telephony by enabling ViPr users to add unicast telephone calls to ViPr conferences.
The UAM adds conferencing capabilities between the ViPr terminals and telephone users (i.e. PSTN, Mobile phones and SIP phones) by converting upstream unicast voice-only telephone streams into point-to-multipoint streams (i.e. PMP-SVC or IP Multicast) and converting downstream ViPr multicast/PMP audio streams to unicast telephone voice-only streams as well as performing downstream audio transcoding of ViPr audio from wideband 16 bit/16 KHz PCM encoding to G.711 or G.722.
The UAM is designed to:                a. Enable point-to-point voice-only calls between one ViPr user and one telephone user.        b. Enable adding one voice-only telephone call to a multi-party ViPr audio/video conference.        c. Perform downstream audio transcoding of ViPr audio streams from the wideband 16 bit/16 KHz PCM to G.711 or G.722.        d. Operate as an Intermedia gateway bridging between ViPr terminals on an ATM network and SIP-based VoIP telephony gateways on Ethernet networks.        